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Phone: 989.837.3790
Toll Free: 888.866.4638
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Kevin Smith
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Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
www.mercury.net
No, even with the numerical IP addresses they still had the problem.
Kevin
Mike wrote:
I`m curious: did going with numerical IP addresses fix your problem?
Mick
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent
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we shall see if this helps.
Thanks,
Kevin
Mike wrote:
I`m curious: did going with numerical IP addresses fix your problem?
Mick
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Wednesday, June 04, 2008 13:10
connection. So far, every one I have tried (Record, Monitor, MixMonitor)
does not seem to create the file. Asterisk version is 1.2.
Thanks,
Kevin
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Kevin Smith
---
Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
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Kevin Smith
Yes, I was using a name instead of an IP address. And if memory
servesI *think* it is using TCPprefered...but I could be wrong.
Kevin
Mike wrote:
I have been running into a few issues with Asterisk/polycom and I am
running out of ideas. This problem has been ongoing for the last couple
JR Richardson wrote:
You mentioned this started happening 3 months ago, what happened then?
Network changes, equipment changes, traffic increased, new users
(downloading allot during the day, surfing porn), wireless
interference?
The initial problem started when our DS3 was throwing
Perhaps seeing some of your dial plan (such as the macro, etc) would
help not only me, but also others, because maybe I am just not following
you.
Off the top of my head there are a few things you could do..but again,
it depends on how your dialplan is set up and how you access the macro.
One
I guess take this into consideration if time isn't a real factor
(however, I'm sure it is). In my experience I found it best to start
learning with the configuration files only then use the GUI. The GUI's
are very nice and handy, but sometimes I feel they lack what you could
do with manually
I almost hate to admit this...but I'm still running Asterisk 1.2 on
Fedora 4 :D
However, I'm planning on upgrading to 1.4 but it has been working out
just fine so far and I just can't find time to upgrade. Otherwise, at
least with Fedora I have had no major issues running Asterisk. Most of
Hi Robert,
While I'm not sure how our network compares with yours, we run about
twenty 601 phones along with our office workstations (some stations are
without a phone). Each station with a phone is connected with the other
Ethernet port on the phone so we have one drop to each station. The
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between
certain hours. I have a few checks in place before a caller enters,
first it checks to see if there it is within the time window, then
checks to see if there are any agents log into queue, if any fail they
get
I'm not sure of a way to do it through AGI, but I know you could make
the script take the recording, use sox to convert it to the file format
you need, then maybe use like a Flash media player to control the
playback of the sound file. It is a bit clunky but it was just one of
the ideas (the
rachid wrote:
Hi,
I have a queue with maxlen=1, and when i make a call, the call enters
into the queue,
but he doesn't exit from it after a reject signal received from the
agent??
please, have you any idea how to remove calls after a reject signal???
Thanks.
Rachid
the db. Something like this:
Application: AGI
Data: inform.php|68456943
Kevin Smith wrote:
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data:
Hi everyone,
Simple question that I haven't been able to find a direct answer to. We
currently have call recording with our asterisk system. The files, I am
assuming since that is the codec we are using, are being recorded in the
g729 codec. Is there a way to listen to these calls, say on
Hi Kevin,
Thanks, that's what I thought but sometimes you need a second opinion
from someone with more experience to get administration off your back
about an issue such as this.
Kevin
Kevin P. Fleming wrote:
Kevin Smith wrote:
We are running Polycom 601's. I can't seem to find
Hi everyone,
We are running Polycom 601's. I can't seem to find anything to say one
way or another on this issue, so I figured I would ask. I have call
waiting notification working on the phones when a user is on the phone.
However, is it possible to see the notification on the screen or hear
There are a few things to look at.
First off, you have a lot of wildcard testing that is probably throwing
the dial plan off. For example, you have the following:
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten =
John Marvin wrote:
Kevin Smith wrote:
Here is what the configuration looks like for one of the phones, the
other is 284:
[283](Empire-Defaults)
[EMAIL PROTECTED]
[283a](Empire-Defaults) [EMAIL PROTECTED]
[283b](Empire-Defaults)
[EMAIL PROTECTED]
So actually you are trying to use
Sorry for the late reply, school has started again so I am not in the
office as much. I also remove all the old postings I didn't need and
also deleted mine, so if there were any before this with questions that
I still haven't answered, let me know.
Rich Adamson wrote:
John Marvin wrote:
Hi everyone,
While this isn't a true asterisk question, I know a lot of people here
use Polycom phones. Anyway, I have two Polycom 601 phones that share the
same voicemail box. Now it is intermittent, but sometimes both phones
will have a notification there is a voice mail, but then sometimes
, at 00:42, Kevin Smith wrote:
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I have
to choose between FreeTDS and unixODBC. Is there a better choice?
Which option would be better in the log run
Hi Mike,
As far as I know, you need to at least start the dialing (ie New call,
speaker, etc) for the digitmap to even come into play.
The only settings that I am aware of that you can try to change are
dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf.
Kevin
Mike
Hi everyone,
I am looking to log CDR records to our MSSQL database for further
examination on the records. From what I gathered from the wiki I have to
choose between FreeTDS and unixODBC. Is there a better choice? Which
option would be better in the log run?
Also configuration asterisk to
Well personally I am just glad I wasn't the only one seeing the
problem. As much as I don't like the place 100% of the blame on
something unless I fully know what is going on, in this case Asterisk,
but I couldn't see any solution but a bug.
Personally I wouldn't mind testing out the
Dialing a number and transferring a number are two different things. And
no offense, you are not really providing a lot of details along with
your problem. So you can dial the numbers but not transfer from one to
the other.
What does the CLI say when you try the transfer? That would provide a
Hi Avi,
I had a similar problem. Have extension 405 put the call on hold (after
the transfer) and then off hold. I am willing to bet it will bring back
the audio stream. I posted something similar a few weeks ago and if
anyone thought it was a bug, to let me know what information I needed to
loop as the first pick.
dbc.
Kevin Smith wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2 =
66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 =
Doug,
Note: Don't take this email serious, I'm just messing with you, but it
sure as poop is ;).
In version 1.6.x released 18th of July 2005 in section 2.2.1.4, Reset
the Factory Defaults
"To perform this function on all phones except the IP4000,
simultaneously press and hold 4,6,8 and *
.
By that I would say placing them on hold clears a flag or updates one to
connect the audio stream? Or am I way off on this assumption? Also if
this sounds like a possible bug, what information do I need to include,
or is good to include, when submitting bugs?
Thanks,
Kevin
Kevin Smith wrote:
Hey
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing
until it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY),
If I am following you right, for extension matching you need to have a
_ in front of the number.
So your example should be like this:
exten = _949927,1,Goto(mainmenu,s,1)
Also I don't know if you did this on purpose or not but N will only
match for numbers 2-9, if you want 0-9 you will
:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want
I'm sure there probably is other ways to do this but you could write a
script as a cron to use the manager API, filter the data you want, and
store it in a database or text file. But depending how often you run it,
you may miss some data.
Douglas Garstang wrote:
Thanks Johann. Yes, I wish
Hi Santiago,
Unless it is a typo on the wiki, I think you want your queue.conf to be
like this:
member = Agent/@1
member = Agent/:2,1
That way you include group 1, and then include group 2 with
consideration of penalty. From the problem you are having it sounds like
the agent whose phone
Hi Julian,
If the 301's support ACD log in and log out, they should display a soft
button showing the current status of the phone, I know for sure the
601's do. Personally with our 601's I used two of the contact lines and
made my own log in and logout buttons and wrote my own script to log
If you turn verbose on under the remote console for asterisk does it
show any information that phones are trying to register or anything for
that matter? Another thing you may want to verify is that the phones
are communicating with the server if you aren't seeing anything on
verbose. I am
Stephen,
I would check with your polycom reseller. They should have the files
you are looking for and you know they will be at least from a
creditable source. In terms of setting up your phones for ftp
provisioning, you will need to edit the files that you obtain from the
reseller, and edit
Michael,
Maybe I am not understanding your question, are you saying that when you
configure your phone with a static IP address, you cannot find the boot
server and when in DHCP you can? If you are having problems with the
phone having a static IP address, make sure it is getting the correct
the road we plan on adding a database for call
logging, configurations, etc, and I would agree with you Jay, storing
the variable there would be the better choice.
Thanks again.
Kevin
Jay Milk wrote:
Kevin Smith wrote:
Hi everyone,
I know that functions like set_variable and get_variable
, you
don't have to redo all the phones with the new address, the DHCP server
will take care of it, but they are a bit more work.
Kevin
Michael Welter wrote:
Kevin Smith wrote:
Michael,
Maybe I am not understanding your question, are you saying that when
you configure your phone
Hi everyone,
I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it
Hi everyone,
Can someone post an example of how you read in a channel variable from
asterisk through PHP. I tried the ones voip-info.org but none of them
seem to work, or at least I am not doing something write, but I have no
problem setting variables and other functions, just reading
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI Script
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but it all started working,
AGI classes and all.
Thanks again,
Kevin
Time Bandit wrote:
I have tried both ways (with PHPAGI
I agree, that's what every example I saw was using. But ya, it's working
now so I'm a happy camper :D
Time Bandit wrote:
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but
We have had this problem too, but just not as frequently as others are
reporting. I started to write a PHP script as a workaround to browse all
of the mail box folders and remove the txt file that is not needed.
However, I haven't tested it fully to make sure it doesn't mess anything
else up.
Hey everyone,
I wrote in last week about our Polycom phones rebooting. I had a nice
theory with it being the PoE switch but that was thrown out the window
today when phones even with a power supply rebooted.
So my question now points back to Asterisk. Is there any feature on
Asterisk that
, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to
be set to 1.
otherwise... beats the heck out of me!
-Original Message-
From: Kevin Smith [mailto:[EMAIL PROTECTED]
Sent: Friday, June 30, 2006 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Auto
: Kevin Smith [mailto:[EMAIL PROTECTED]
Sent: Friday, June 30, 2006 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Auto NOTIFY
Hey Doug,
That's what I figured, but correct me if I am wrong. Isn't 1
will always
set the phones to reboot on a NOTIFY
Hey everyone,
I know this isn't a direct Asterisk issue, but some of you may know this
answer.
I recently upgraded the SIP version to 1.6.6 on all of our phones in the
office. Everything is working fine, except one aspect. The phones in the
office reboot randomly for no apparent reason. I
if it did.
Thanks for the idea Chris.
Kevin
Chris Mason (Lists) wrote:
Kevin Smith wrote:
Any other thoughts as to what may have caused the phone to reboot?
the power supplies on these phones are very underrated and any power
fluctuation will cause them to reboot. I get it when we
I'm not to familiar with Express Talk, but try removing the username=200
from your sip definition. From your lines menu it doesn't look like you
are sending a username to asterisk. The SIP number is probably going
reference to the sip context and since you are telling asterisk there is
a
Hey everyone,
Hopefully this will be simple enough to answer. I have a menu setup like
below:
exten = 850,n,Set(MenuLoop=1)
exten = 850,n,Playback(mercury-prompts/welcome)
exten =
850,n(MainMenu),Background(mercury-prompts/MainMenu-if-you-know-the-ext)
exten =
Hi Patrick,
Let me see if I am following you here. When a caller calls in, obviously
you want them to be in the first queue level based on your dial plan.
Now, how do you want the caller to reach the next queue? Is the only way
a caller going to go to the next queue via a transfer from the
Hey Doug,
Few things you can do. First off, are the numbers for incoming callers
or for when you are making a call? One way that we do it because our
numbers change a lot is I have a text file with all the numbers on it.
Like below:
[localtoolexchange]
exten = _342, 1, Goto(whereever)
etc..
Hi Stephen,
I use the 601's but I don't think they are THAT much different that
this information won't be helpful or get you in the right direction.
What is your network setup like? Are you using NAT or does the phone
have a public IP address? Also are you seeing any errors on the CLI of
Hi Attilla,
I'm not sure if there is something like that available or not, but I
know there are some alternatives. You can set the time out limit to say
15 seconds, which for me is about 3-4 rings on the phone before it goes
looking for the next agent. The other option you can manually remove
Hi Stephen,
Sorry if the e-mail is a bit choppy but I figured it would be best to
cut/paste answers in. Now again, I am using the 601's so things may be a
little different, but for the most part should be similar.
No NAT. This is just one Polycom 501 that is dialing out through an
Asterisk
Hey everyone,
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which
in users. Is there a way to find out
which extensions are currently logged in??
Thanks agai
On 4/24/06, *Kevin Smith* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
What I would suggest doing, since we have a similar setup (where
our 24
support contracts can enter a pin
Hey everyone,
Hopefully someone can point me in the right direction for this.
Currently we have two offices, all using Polycom 601 Revsion E I think.
All have the same configurations and firmware versions.
The differences:
Office A: public IP address.
Office B: NAT (router has a static IP)
Hi,
What I would suggest doing, since we have a similar setup (where our 24
support contracts can enter a pin number to be routed to an on call
tech) is I use the function QUEUEAGENTCOUNT(queue name). Since you said
that the calls should only be routed after the last support person logs
out,
Hey everyone,
Hopefully I can describe the problem well enough so bear with me.
There are 3 companies that are tied into our asterisk server. Company A
(us) uses the default settings for music on hold. Companies B and C
however, want something different. For them I have when a call comes
Hi everyone,
On the Polycom 601 phones we are using, the forward feature works very
nicely for agents that are out on trips. I was wondering if there is a
way to test to see if they have the forward option enabled.
When it is enabled the call comes in and gets -- Got SIP response 302
Moved
Hi everyone,
I have been having some problems lately with our PRI and Asterisk, or
maybe it is just me. It happens once maybe twice a day, but when some of
our customers are calling in, the phone just drops on them. I pulled the
information below from the log from one that happened. I notice
that after I sent the e-mail. I was using asterisk for testing before we
hooked the PRI in and it was one of those overlooked items.
Thanks,
Kevin
Michael Welter wrote:
Post your 'cat /proc/interrupts' for us.
Kevin Smith wrote:
Hi everyone,
I have been having some problems lately with our PRI
Hi Matt,
We have somewhat of a similar setup here in my office. We have multiple
queues to which different agents are a member to anyone of them.
Basically what I chose to do was make my own custom log in script. I
reference to the voicemail box and use the ID and password to
authenticate
As far as I can tell everything is pretty much the same. Below is the
debug output for a particular phone I left a voicemail for. Maybe I am
missing something that I am just not seeing. Otherwise I'm still not
getting a count, but the other notifications are still working.
Thanks again,
Kevin
Hey William,
Yes, Mercury-Network-Emp is the context of my voicemail.conf, which is
why in the sip it has the @Mercury-Network-Emp so it knows which context
to apply it to. Any other ideas?
Thanks,
Kevin
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While I know this is not a true asterisk problem, I figure someone where
may know. When you click on Messages and it gives you the count of
Urgent, New, etc. How can you make the phone gather that information?
For example, my phone shows me there is an e-mail. It also sends an
e-mail. Yet,
voicemail on the same server that the phone
registers with? As long as your mwi is working, it should
automatically receive a count of how many messages you have from
asterisk.
Aaron
On Fri, 24 Mar 2006, Kevin Smith wrote:
While I know this is not a true asterisk problem, I figure someone
where
Hey everyone,
I have been trying to figure this out and I am just getting no where
with it. The office is using Polycom IP 601 phones. Everything sounds
great in terms of quality on both heads. However, users of the phone are
having trouble with their headsets and handsets. Some users are
Do you mean, say number 444-555- calls in. You want to hit dial for
that number, from say the missed calls list, and have it on add a 9 in
front? If so just do this in extensions.conf
exten = _9NXXNXX,1,Dial(Zap/g1/${EXTEN} ;Takes calls with a 9
exten =
Hey everyone,
I have noticed a few questions close to the issue I am having but I
haven't seen any that quite match the problem I am seeing.
I have 3 queues. Some members share one queue and some are completely
separate. Some members have a higher penalty then others. I am using
Hey everyone,
We have a special mail box for certain customers when we are out of the
office. Basically they enter a pin number and if it is valid they leave
a message and it notifies the on call techs. My question is regarding
externnotify for the voice mail.conf. If I enabled that and set
Hey everyone,
I know this is a problem with mpg123, but it just started happening and
I have no idea why. I haven't changed any of the audio format settings
yet. Before tonight, I was able to call, listen to the queues, hear the
music on hold, no problems. I added a new context to a dial
Hey Rich and everyone.
I tried what you suggested, and it didn't work. I even recomplied
everything, moved all of my configuration files out and remade the
samples, so as far as I can tell everything is back to day 1. However,
it is still pulling in the database information. This is really
Hey everyone, I haven't seen an issue quite like mine, so I am hoping
anyone who used the Polycom 601's may have an idea.
We are going to be switching our office over to Asterisk. All the phones
are going to be 601's, I am going to set up a boot server, but for now I
am just going to test
C F wrote:
What does the dialplan for the Polyocm 601 (the one the phone uses,
not Asterisk) look like?
You can see if it's a polycom or asterisk thing, by enabling sip
debug, and watch what is coming in from the Polycom. if nothing is
coming then it's the Polycom doing it.
On 2/23/06, Kevin
Hey everyone,
I have a more of an opinion question then a technical question. The
asterisk server I am setting up is going to host 3 different businesses.
Each business is in the same building, and on the same network. My
question is regarding calls coming in and going out. We are a small ISP
Hey everyone,
I am having a little trouble getting this section of the dial plan
configured. Does anyone know of a way I can get the number of agents
that are currently logged into a queue? My goal is if no agent is logged
in the queue, it gives customers the message we are closed depending
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